Hello world, I've got two FS servers configured as gateways for each other and I'm currently testing the telephony. Usinge the ILBC audio codec, I figured out that one of the FS servers doesn't forward RTP streams correctly to the other server. Here is its status-quo: INPUT = proper ILBC payload type (97 or 108) OUTPUT = unknown payload type (97 or 102) I've already changed the parameters in internal.xml & external.xml: <param name="inbound-codec-negotiation" value="greedy"/> <param name="disable-transcoding" value="true"/> When dialing out, I also use the following syntax:{absolute_codec_string='GSM,PCMU'}sofia/gateway/mygateway/mynumber Is there another thing to do to have proper ILBC streams passing through the gateways ? Thanking y'all in advance ;) BR, David N. Hello world, I've got two FS servers configured as gateways for each other and I'm currently testing the telephony. Usinge the ILBC audio codec, I figured out that one of the FS servers doesn't forward RTP streams correctly to the other server. Here is its status-quo: INPUT = proper ILBC payload type (97 or 108) OUTPUT = unknown payload type (97 or 102) I've already changed the parameters in internal.xml and also in external.xml: <param name="inbound-codec-negotiation" value="greedy"/> <param name="disable-transcoding" value="true"/> Is there another thing to do to have proper ILBC streams passing through the gateways ? Thanking y'all in advance ;) BR, David N.